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Posted to user@openmeetings.apache.org by Maxim Solodovnik <so...@gmail.com> on 2013/02/01 06:00:28 UTC
Re: SIP connectivity
I have updated the instruction (minor update)
On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> OK will add it and notify you
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>> wrote:
>>
>>> It is for Asterisk 11 - don't know for other versions. You probably
>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>> Asterisk source should be compared across versions.
>>>
>>> this one is missing:
>>>
>>> `useragent` varchar(20) DEFAULT NULL,
>>>
>>> complete list (I think) is on:
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>
>>>
>>> If I bump into others, I'll report ASAP,
>>>
>>>
>>> BC
>>>
>>>
>>>
>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll add it.
>>> My purpose was to create table with required fields only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <bart.coninckx@telenet.be
>>> > wrote:
>>>
>>>> Openmeetings installed them for me, that's why I ended up with those.
>>>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>>>> have 'em removed from the install procedure.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> If you look in the source directory of your asterisk tar file, under
>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the ones with OM.
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I obviously used the one in the
>>>> openmeetings mysql database, but this one seems to miss the table
>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> From an asterisk configuration standpoint there are very few
>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>> changes that I ran into (in my production environment) was changes to SIP
>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>> skim through the change log for full details, but I think that was the jist
>>>> of it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>> desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> That is amazing - I initially tried to do the same thing by using the
>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>> and has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>
>>>>
>>>>
>>>> We are currently testing it and trying to add video capabilities ...
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>> just a bunch of command line instructions are given.
>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>> meetme database" actually says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>> this is the one you're already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>>> a MeetMe conference? Or is it the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential information and is intended only
>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>> are not the intended recipient of this email, you are hereby notified that
>>>> any unauthorized use, dissemination or copying of this email or the
>>>> information contained in it or attached to it is strictly prohibited. If
>>>> you received this message in error, please immediately notify the sender at
>>>> Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
We already set two columns: name and defaultuser:
https://svn.apache.org/repos/asf/openmeetings/trunk/singlewebapp/src/org/apache/openmeetings/persistence/beans/sip/asterisk/AsteriskSipUser.java
On Thu, Feb 7, 2013 at 10:07 PM, Bakko <as...@gmail.com> wrote:
> Other think,
>
> On the openmeetings table sipuser, field username is deprecated. Change to
> defaultuser.
>
> Regards
>
> El 07/02/2013 09:59, Maxim Solodovnik escribió:
>
> Just have tested it and it it works!
> Thanks a lot!
> I'll update the documentation
>
>
> On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
>
>> Hello,
>>
>> on Asterisk 1.8, extconfig.conf this line:
>>
>> *sipusers => odbc,asterisk,sipusers
>>
>> *is deprecated.
>>
>> Now only use:
>>
>> *sippeers => odbc,asterisk,sipusers
>>
>> *Regards*
>>
>> *
>> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>>
>> I have updated the instruction (minor update)
>>
>>
>> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> Hello Bart,
>>>
>>> I just take a look at your URL ...
>>> OM does not create/use sipfriends DB table (at least from version 2.1)
>>> only meetme table is used
>>>
>>> so I'm afraid there is nothing to change here
>>>
>>> Here is the most recent instruction:
>>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>>
>>> Will ask our SIP guru to review it one more time :)
>>>
>>>
>>>
>>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>>
>>>> OK will add it and notify you
>>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>>> wrote:
>>>>
>>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>>> Asterisk source should be compared across versions.
>>>>>
>>>>> this one is missing:
>>>>>
>>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>>
>>>>> complete list (I think) is on:
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>
>>>>>
>>>>> If I bump into others, I'll report ASAP,
>>>>>
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>>
>>>>> Is the OM meetme table incomplete?
>>>>> My asterisk reports no issues :(
>>>>>
>>>>> could you provide me with missing fields and I'll add it.
>>>>> My purpose was to create table with required fields only.
>>>>>
>>>>>
>>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>>> idea to have 'em removed from the install procedure.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> If you look in the source directory of your asterisk tar file, under
>>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>>> *To:* user@openmeetings.apache.org
>>>>>> *Cc:* Jeff Clay
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> Well,
>>>>>>
>>>>>> I might have found one difference though:
>>>>>>
>>>>>>
>>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>> dictates how the table should look like. I obviously used the one in the
>>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> From an asterisk configuration standpoint there are very few
>>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>>> skim through the change log for full details, but I think that was the jist
>>>>>> of it.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>>> *To:* Maxim Solodovnik
>>>>>> *Cc:* user
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>>> desired.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>>
>>>>>> I test the integration using
>>>>>>
>>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>>
>>>>>> Are you guys using Asterisk 11? This version is the newest LTS
>>>>>> version and has the best video capabilities.
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>>
>>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>>
>>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>>
>>>>>>
>>>>>>
>>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> Hi Jeff,
>>>>>>
>>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>>> just a bunch of command line instructions are given.
>>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>>> meetme database" actually says it all in one go :-)
>>>>>>
>>>>>> cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>>> database. Have you read this page
>>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>>> this is the one you're already referring to.
>>>>>>
>>>>>> Jeff Clay
>>>>>> Network Administrator
>>>>>> Infotech Enterprises America
>>>>>> 870-215-5506
>>>>>> Ext. 1506
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>>> To: user@openmeetings.apache.org
>>>>>> Subject: SIP connectivity
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>>
>>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>>
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> Bc
>>>>>>
>>>>>> ________________________________
>>>>>>
>>>>>> DISCLAIMER:
>>>>>>
>>>>>> This email may contain confidential information and is intended only
>>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>>> you received this message in error, please immediately notify the sender at
>>>>>> Infotech and delete the original message.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bakko <as...@gmail.com>.
Other think,
On the openmeetings table sipuser, field username is deprecated. Change
to defaultuser.
Regards
El 07/02/2013 09:59, Maxim Solodovnik escribió:
> Just have tested it and it it works!
> Thanks a lot!
> I'll update the documentation
>
>
> On Fri, Feb 1, 2013 at 7:35 PM, Bakko <asannucci@gmail.com
> <ma...@gmail.com>> wrote:
>
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> /sipusers => odbc,asterisk,sipusers
>
> /is deprecated.
>
> Now only use:
>
> /sippeers => odbc,asterisk,sipusers
>
> /Regards/
>
> /
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>> I have updated the instruction (minor update)
>>
>>
>> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik
>> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from
>> version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
>> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>>
>> OK will add it and notify you
>>
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> It is for Asterisk 11 - don't know for other
>> versions. You probably have no issues because of the
>> 1.8 version. To be sure the .sql files in the
>> Asterisk source should be compared across versions.
>>
>> this one is missing:
>>
>> `useragent` varchar(20) DEFAULT NULL,
>>
>> complete list (I think) is on:
>>
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>
>>
>> If I bump into others, I'll report ASAP,
>>
>>
>> BC
>>
>>
>>
>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll
>>> add it.
>>> My purpose was to create table with required fields
>>> only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>>> <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Openmeetings installed them for me, that's why I
>>> ended up with those. Using the Asterisk ones
>>> makes more sense to me. Maybe it's a good idea
>>> to have 'em removed from the install procedure.
>>>
>>> BC
>>>
>>>
>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> If you look in the source directory of your
>>>> asterisk tar file, under contrib/realtime/mysql
>>>> you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the
>>>> ones with OM.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I
>>>> obviously used the one in the openmeetings
>>>> mysql database, but this one seems to miss the
>>>> table "useragent". I discovered this because it
>>>> showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> From an asterisk configuration standpoint
>>>> there are very few differences between
>>>> 1.8.x and 11.x. If memory serves, the only
>>>> major changes that I ran into (in my
>>>> production environment) was changes to SIP
>>>> NAT values and the behavior of app_page()
>>>> now uses confbridge instead of meetme to
>>>> mix the audio. Also, TCP, TLS and
>>>> app_confbridge got a major overhauling.
>>>> There were of course many other changes and
>>>> bug fixes, you can skim through the change
>>>> log for full details, but I think that was
>>>> the jist of it.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> I see - I'm willing to try the 11 version
>>>> in the next fiew days if desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart
>>>> Coninckx <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>> wrote:
>>>>
>>>> That is amazing - I initially tried to
>>>> do the same thing by using the new
>>>> chan_motif driver in Asterisk 11 which
>>>> connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This
>>>> version is the newest LTS version and
>>>> has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user
>>>> in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM
>>>> room using SIP hard or soft phone.
>>>>
>>>> We are currently testing it and
>>>> trying to add video capabilities ...
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM,
>>>> Bart Coninckx
>>>> <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>>
>>>> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none
>>>> explain what they set up to do,
>>>> just a bunch of command line
>>>> instructions are given.
>>>> Your "OM will create a meetme
>>>> meeting as configured in the
>>>> realtime meetme database" actually
>>>> says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as
>>>> configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>>> ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html
>>>> but I assume this is the one you're
>>>> already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be
>>>> <ma...@telenet.be>]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how
>>>> to connect OM with a SIP proxy or
>>>> server, more particularly with the
>>>> MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not
>>>> mentionned however. Will OM callout
>>>> to a MeetMe conference? Or is it
>>>> the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential
>>>> information and is intended only
>>>> for the use of the specific
>>>> individual(s) to which it is
>>>> addressed. If you are not the
>>>> intended recipient of this email,
>>>> you are hereby notified that any
>>>> unauthorized use, dissemination or
>>>> copying of this email or the
>>>> information contained in it or
>>>> attached to it is strictly
>>>> prohibited. If you received this
>>>> message in error, please
>>>> immediately notify the sender at
>>>> Infotech and delete the original
>>>> message.
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Just have tested it and it it works!
Thanks a lot!
I'll update the documentation
On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> *sipusers => odbc,asterisk,sipusers
>
> *is deprecated.
>
> Now only use:
>
> *sippeers => odbc,asterisk,sipusers
>
> *Regards*
>
> *
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Thanks!
I'll try to test it and modify the doc.
(I leave it since I'm not "asterisk guru" and try not to experiment :)) )
On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> *sipusers => odbc,asterisk,sipusers
>
> *is deprecated.
>
> Now only use:
>
> *sippeers => odbc,asterisk,sipusers
>
> *Regards*
>
> *
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bakko <as...@gmail.com>.
Hello,
on Asterisk 1.8, extconfig.conf this line:
/sipusers => odbc,asterisk,sipusers
/is deprecated.
Now only use:
/sippeers => odbc,asterisk,sipusers
/Regards/
/
El 01/02/2013 00:00, Maxim Solodovnik escribió:
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> OK will add it and notify you
>
> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
> <bart.coninckx@telenet.be <ma...@telenet.be>>
> wrote:
>
> It is for Asterisk 11 - don't know for other versions. You
> probably have no issues because of the 1.8 version. To be
> sure the .sql files in the Asterisk source should be
> compared across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>> Is the OM meetme table incomplete?
>> My asterisk reports no issues :(
>>
>> could you provide me with missing fields and I'll add it.
>> My purpose was to create table with required fields only.
>>
>>
>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> Openmeetings installed them for me, that's why I
>> ended up with those. Using the Asterisk ones makes
>> more sense to me. Maybe it's a good idea to have 'em
>> removed from the install procedure.
>>
>> BC
>>
>>
>> On 01/30/13 22:30, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> If you look in the source directory of your asterisk
>>> tar file, under contrib/realtime/mysql you’ll find
>>> the .sql files required for all the realtime
>>> drivers. I never thought to use the ones with OM.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>> *To:* user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> *Cc:* Jeff Clay
>>> *Subject:* Re: SIP connectivity
>>>
>>> Well,
>>>
>>> I might have found one difference though:
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>> dictates how the table should look like. I obviously
>>> used the one in the openmeetings mysql database, but
>>> this one seems to miss the table "useragent". I
>>> discovered this because it showed up in the logfiles.
>>>
>>> BC
>>>
>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> From an asterisk configuration standpoint there
>>> are very few differences between 1.8.x and 11.x.
>>> If memory serves, the only major changes that I
>>> ran into (in my production environment) was
>>> changes to SIP NAT values and the behavior of
>>> app_page() now uses confbridge instead of meetme
>>> to mix the audio. Also, TCP, TLS and
>>> app_confbridge got a major overhauling. There
>>> were of course many other changes and bug fixes,
>>> you can skim through the change log for full
>>> details, but I think that was the jist of it.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>> *To:* Maxim Solodovnik
>>> *Cc:* user
>>> *Subject:* Re: SIP connectivity
>>>
>>> I see - I'm willing to try the 11 version in the
>>> next fiew days if desired.
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>
>>> I test the integration using
>>>
>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>
>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> That is amazing - I initially tried to do
>>> the same thing by using the new chan_motif
>>> driver in Asterisk 11 which connects to a
>>> XMPP server.
>>>
>>> Are you guys using Asterisk 11? This version
>>> is the newest LTS version and has the best
>>> video capabilities.
>>>
>>> Cheers,
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>
>>> red5sip will create special OM user in
>>> the room: "SIP Transport"
>>>
>>> after that you can call to the OM room
>>> using SIP hard or soft phone.
>>>
>>> We are currently testing it and trying
>>> to add video capabilities ...
>>>
>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Hi Jeff,
>>>
>>> In fact, I saw both pages, but none
>>> explain what they set up to do, just a
>>> bunch of command line instructions are
>>> given.
>>> Your "OM will create a meetme meeting as
>>> configured in the realtime meetme
>>> database" actually says it all in one go
>>> :-)
>>>
>>> cheers,
>>>
>>> BC
>>>
>>>
>>>
>>>
>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as
>>> configured in the realtime meetme
>>> database. Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>> ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html
>>> but I assume this is the one you're
>>> already referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be
>>> <ma...@telenet.be>]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to
>>> connect OM with a SIP proxy or server,
>>> more particularly with the MeetMe
>>> application in Asterisk.
>>>
>>> The exact goal or purpose is not
>>> mentionned however. Will OM callout to a
>>> MeetMe conference? Or is it the other
>>> way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
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>>>
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>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax